The present invention is related to the field of signal transmission in the local access loop between a central telephone office switching center and a plurality of local users. In particular, a packetized digital network architecture is provided that is capable of transporting high-quality voice and high-speed data packets between the central office and the plurality of users. This data-centric architecture provides many advantages over traditional voice-centric networks presently in use.
Prior to the explosive growth in the public's demand for data services, such as dial-up Internet access, the local loop access network transported mostly voice information. This present access network typically includes numerous twisted-pair wire connections between the plurality of user locations and a central office switch. These connections can be multiplexed in order to more efficiently transport voice calls to and from the central office. The present access network for the local loop is designed primarily to carry these voice signals, i.e., it is a voice-centric network.
Today, data traffic carried across telephone networks is growing exponentially, and by many measures may have already surpassed traditional voice traffic, due in large measure to the explosive growth of dial-up data connections. The basic problem with transporting data traffic over this voice-centric network, and in particular the local loop access part of the network, is that it is optimized for voice traffic, not data. The voice-centric structure of the access network limits the ability to receive and transmit high-speed data signals along with traditional quality voice signals. Simply put, the access part of the network is not well matched to the type of information it is now primarily transporting. As users demand higher and higher data transmission capabilities, the inefficiencies of the present access network will cause user demand to shift to other mediums of transport for fulfillment, such as satellite transmission, cable distribution, wireless services, etc.
An alternative present local access network that is available in some areas is a digital loop carrier (“DLC”) system. DLC systems utilize fiber-optic distribution links and remote multiplexing devices to deliver voice and data signals to and from the local users. DLC systems are synchronous networks that include a device known as a Time-Slot Interchanger (“TSI”). The TSI allocates the available bandwidth of the DLC system in “chunks,” and maps incoming DS-0 PCM telephone circuits from a digital switch to the allocated chucks of bandwidth. Each DS-0 telephone line is a digital 64 Kbps PCM-modulated link. (A DS-1 line comprises 24 DS-0 lines.)
The utilization of system bandwidth in a DLC network is non-optimal because the TSI assigns and maps a particular number of DS-0 lines to the available bandwidth of the system, whether or not those lines are being actively used for voice information. The relatively constant mapping function of the TSI creates “stranded bandwidth,” i.e., bandwidth that is not being used by the system, and which cannot be reallocated to other links or users. By mapping the bandwidth in chunks to particular DS-0 lines, regardless of use, the TSI inevitably sets aside bandwidth that is unused by the system. Thus the TSI is non-optimal. In addition to this stranded bandwidth problem, the presently available DLC systems are complex, costly, and do not scale very effectively, meaning that it is not easy to expand the DLC system once it has been implemented in a particular area.
Another method of transporting voice and data in the local access loop is via a dial-up TCP/IP connection to the Internet. The dial-up connection to the Internet is created using a computer modem connection to a local Internet Service Provider (“ISP”) over the standard voice-centric access network. This technique layers a digital data packet protocol (TCP/IP) on top of the analog voice circuit connecting the central office switch and the local user. Data signals are transported as TCP/IP packets at speeds of 30–50 Kbps, assuming a standard 56 Kbps modem is utilized. Voice signals can also be transported over this packet connection using a technique known as IP Telephony.
IP Telephony is a software transport technique that digitizes the user's voice, compresses the digitized voice signals, and then packs the compressed digitized voice signals into TCP/IP packets for transport across the dial-up connection. The main problems with IP Telephony are processing overhead; poor sound quality; and packet delay. Because of the asynchronous nature of the TCP/IP connection to the Internet, voice packets can get lost or delayed in transit, leading to a garbled sounding voice signal. This is unacceptable for most telephone customers. Furthermore, because of the need to process and compress the speech signals, IP Telephony adds significantly delays to the voice connection. This further erodes the quality of the voice signal. In addition, it is presently not possible to operate a fax or modem connection over an IP Telephony link, which further limits its general applicability to the local access loop.
In summary, none of the presently available techniques for transporting voice and data signals in the local access loop are optimized for transporting both voice and data traffic. Therefore, there remains a general need in this art for a network architecture for simultaneously transporting high-quality voice and high-speed data signals in the local access loop.
There remains an additional need for such a network architecture that is cost effective, scalable, bandwidth efficient, and is designed to evolve (or scale) as advances are made in digital packet switching hardware.
There remains an additional need for such a architecture in which voice and data signals are packetized and transported in the local access loop using packet-switching hardware that is readily available, highly integrated and cost effective.
There remains yet an additional need for a packet-switched local loop access system for transporting voice and data packets in which the voice packets are prioritized in order to ensure quality sound delivery.
There remains yet another need for a method of time-synchronizing the voice packets in such a packet-switched local loop access system.